@decibel83 wrote:
Hi,
I configured a Grandstream HT503 as FXO gateway on FreePBX 14 with Asterisk 14 following the guide at https://wiki.freepbx.org/pages/viewpage.action?pageId=33293313 and configuring the trunk on FreePBX.After configuring the HT503 device and my FreePBX system I can make outgoing calls but if I try to make an incoming call from my mobile phone I hear the ringtones, but the configured extension for the Incoming Route does not ring, and I get these errors into the Asterisk console:
[2017-10-10 19:50:05] NOTICE[14511]: res_pjsip/pjsip_distributor.c:649 log_failed_request: Request 'INVITE' from '<sip:[my_mobile_number]@192.168.61.1>' failed for '192.168.61.11:5160' (callid: 1092213300-5160-4@BJC.BGI.GB.BB) - No matching endpoint found [2017-10-10 19:50:05] NOTICE[14511]: res_pjsip/pjsip_distributor.c:649 log_failed_request: Request 'INVITE' from '<sip:[my_mobile_number]@192.168.61.1>' failed for '192.168.61.11:5160' (callid: 1092213300-5160-4@BJC.BGI.GB.BB) - No matching endpoint found [2017-10-10 19:50:05] NOTICE[14511]: res_pjsip/pjsip_distributor.c:649 log_failed_request: Request 'INVITE' from '<sip:[my_mobile_number]@192.168.61.1>' failed for '192.168.61.11:5160' (callid: 1092213300-5160-4@BJC.BGI.GB.BB) - Failed to authenticateI configured a chan_sip trunk for the HT503, but it seems that the HT503 is using a chan_pjsip trunk (or there should be something not clear to me).
My FreePBX system is using the default ports (so port 5160 for chan_sip and 5060 for chan_pjsip) and I configured the port 5160 for the "Unconditional Call Forward to VOIP / Sip Destination Port" into the basic settings of the HT503.I added a new Incoming route in FreePBX as described on the wiki page, but it seems that it is not called when I make an incoming call.
Could you help me please?
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